X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavformat%2Frtpenc.c;h=d33060770e1d74da3ab3570e2eac0acb7fa340ff;hb=c3386bd5b4d3662f94e902a0fe3e9e869e29967d;hp=c1e8e16f0735e1b3f267c050d80a7736ca176409;hpb=bde15e74de442cebf61acc6cd598a13aca249030;p=ffmpeg diff --git a/libavformat/rtpenc.c b/libavformat/rtpenc.c index c1e8e16f073..d33060770e1 100644 --- a/libavformat/rtpenc.c +++ b/libavformat/rtpenc.c @@ -1,182 +1,338 @@ /* * RTP output format - * Copyright (c) 2002 Fabrice Bellard. + * Copyright (c) 2002 Fabrice Bellard * - * This file is part of FFmpeg. + * This file is part of Libav. * - * FFmpeg is free software; you can redistribute it and/or + * Libav is free software; you can redistribute it and/or * modify it under the terms of the GNU Lesser General Public * License as published by the Free Software Foundation; either * version 2.1 of the License, or (at your option) any later version. * - * FFmpeg is distributed in the hope that it will be useful, + * Libav is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Lesser General Public License for more details. * * You should have received a copy of the GNU Lesser General Public - * License along with FFmpeg; if not, write to the Free Software + * License along with Libav; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ -#include "libavcodec/bitstream.h" #include "avformat.h" #include "mpegts.h" +#include "internal.h" +#include "libavutil/mathematics.h" +#include "libavutil/random_seed.h" +#include "libavutil/opt.h" + +#include "rtpenc.h" + +static const AVOption options[] = { + FF_RTP_FLAG_OPTS(RTPMuxContext, flags), + { "payload_type", "Specify RTP payload type", offsetof(RTPMuxContext, payload_type), AV_OPT_TYPE_INT, {.i64 = -1 }, -1, 127, AV_OPT_FLAG_ENCODING_PARAM }, + { "ssrc", "Stream identifier", offsetof(RTPMuxContext, ssrc), AV_OPT_TYPE_INT, { .i64 = 0 }, INT_MIN, INT_MAX, AV_OPT_FLAG_ENCODING_PARAM }, + { "cname", "CNAME to include in RTCP SR packets", offsetof(RTPMuxContext, cname), AV_OPT_TYPE_STRING, { .str = NULL }, 0, 0, AV_OPT_FLAG_ENCODING_PARAM }, + { "seq", "Starting sequence number", offsetof(RTPMuxContext, seq), AV_OPT_TYPE_INT, { .i64 = -1 }, -1, 65535, AV_OPT_FLAG_ENCODING_PARAM }, + { NULL }, +}; -#include -#include "network.h" - -#include "rtp_internal.h" -#include "rtp_mpv.h" -#include "rtp_aac.h" -#include "rtp_h264.h" - -//#define DEBUG +static const AVClass rtp_muxer_class = { + .class_name = "RTP muxer", + .item_name = av_default_item_name, + .option = options, + .version = LIBAVUTIL_VERSION_INT, +}; #define RTCP_SR_SIZE 28 -#define NTP_OFFSET 2208988800ULL -#define NTP_OFFSET_US (NTP_OFFSET * 1000000ULL) -static uint64_t ntp_time(void) +static int is_supported(enum AVCodecID id) { - return (av_gettime() / 1000) * 1000 + NTP_OFFSET_US; + switch(id) { + case AV_CODEC_ID_H263: + case AV_CODEC_ID_H263P: + case AV_CODEC_ID_H264: + case AV_CODEC_ID_MPEG1VIDEO: + case AV_CODEC_ID_MPEG2VIDEO: + case AV_CODEC_ID_MPEG4: + case AV_CODEC_ID_AAC: + case AV_CODEC_ID_MP2: + case AV_CODEC_ID_MP3: + case AV_CODEC_ID_PCM_ALAW: + case AV_CODEC_ID_PCM_MULAW: + case AV_CODEC_ID_PCM_S8: + case AV_CODEC_ID_PCM_S16BE: + case AV_CODEC_ID_PCM_S16LE: + case AV_CODEC_ID_PCM_U16BE: + case AV_CODEC_ID_PCM_U16LE: + case AV_CODEC_ID_PCM_U8: + case AV_CODEC_ID_MPEG2TS: + case AV_CODEC_ID_AMR_NB: + case AV_CODEC_ID_AMR_WB: + case AV_CODEC_ID_VORBIS: + case AV_CODEC_ID_THEORA: + case AV_CODEC_ID_VP8: + case AV_CODEC_ID_ADPCM_G722: + case AV_CODEC_ID_ADPCM_G726: + case AV_CODEC_ID_ILBC: + case AV_CODEC_ID_MJPEG: + case AV_CODEC_ID_SPEEX: + case AV_CODEC_ID_OPUS: + return 1; + default: + return 0; + } } static int rtp_write_header(AVFormatContext *s1) { - RTPDemuxContext *s = s1->priv_data; - int payload_type, max_packet_size, n; + RTPMuxContext *s = s1->priv_data; + int n; AVStream *st; - if (s1->nb_streams != 1) - return -1; + if (s1->nb_streams != 1) { + av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n"); + return AVERROR(EINVAL); + } st = s1->streams[0]; + if (!is_supported(st->codec->codec_id)) { + av_log(s1, AV_LOG_ERROR, "Unsupported codec %x\n", st->codec->codec_id); + + return -1; + } + + if (s->payload_type < 0) { + /* Re-validate non-dynamic payload types */ + if (st->id < RTP_PT_PRIVATE) + st->id = ff_rtp_get_payload_type(s1, st->codec, -1); - payload_type = rtp_get_payload_type(st->codec); - if (payload_type < 0) - payload_type = RTP_PT_PRIVATE; /* private payload type */ - s->payload_type = payload_type; + s->payload_type = st->id; + } else { + /* private option takes priority */ + st->id = s->payload_type; + } -// following 2 FIXMEs could be set based on the current time, there is normally no info leak, as RTP will likely be transmitted immediately - s->base_timestamp = 0; /* FIXME: was random(), what should this be? */ + s->base_timestamp = av_get_random_seed(); s->timestamp = s->base_timestamp; s->cur_timestamp = 0; - s->ssrc = 0; /* FIXME: was random(), what should this be? */ + if (!s->ssrc) + s->ssrc = av_get_random_seed(); s->first_packet = 1; - s->first_rtcp_ntp_time = AV_NOPTS_VALUE; - - max_packet_size = url_fget_max_packet_size(s1->pb); - if (max_packet_size <= 12) + s->first_rtcp_ntp_time = ff_ntp_time(); + if (s1->start_time_realtime) + /* Round the NTP time to whole milliseconds. */ + s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 + + NTP_OFFSET_US; + // Pick a random sequence start number, but in the lower end of the + // available range, so that any wraparound doesn't happen immediately. + // (Immediate wraparound would be an issue for SRTP.) + if (s->seq < 0) + s->seq = av_get_random_seed() & 0x0fff; + else + s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval + + if (s1->packet_size) { + if (s1->pb->max_packet_size) + s1->packet_size = FFMIN(s1->packet_size, + s1->pb->max_packet_size); + } else + s1->packet_size = s1->pb->max_packet_size; + if (s1->packet_size <= 12) { + av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size); return AVERROR(EIO); - s->max_payload_size = max_packet_size - 12; + } + s->buf = av_malloc(s1->packet_size); + if (s->buf == NULL) { + return AVERROR(ENOMEM); + } + s->max_payload_size = s1->packet_size - 12; s->max_frames_per_packet = 0; - if (s1->max_delay) { - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { - if (st->codec->frame_size == 0) { + if (s1->max_delay > 0) { + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { + int frame_size = av_get_audio_frame_duration(st->codec, 0); + if (!frame_size) + frame_size = st->codec->frame_size; + if (frame_size == 0) { av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n"); } else { - s->max_frames_per_packet = av_rescale_rnd(s1->max_delay, st->codec->sample_rate, AV_TIME_BASE * st->codec->frame_size, AV_ROUND_DOWN); + s->max_frames_per_packet = + av_rescale_q_rnd(s1->max_delay, + AV_TIME_BASE_Q, + (AVRational){ frame_size, st->codec->sample_rate }, + AV_ROUND_DOWN); } } - if (st->codec->codec_type == CODEC_TYPE_VIDEO) { + if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) { /* FIXME: We should round down here... */ - s->max_frames_per_packet = av_rescale_q(s1->max_delay, AV_TIME_BASE_Q, st->codec->time_base); + s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base); } } - av_set_pts_info(st, 32, 1, 90000); + avpriv_set_pts_info(st, 32, 1, 90000); switch(st->codec->codec_id) { - case CODEC_ID_MP2: - case CODEC_ID_MP3: + case AV_CODEC_ID_MP2: + case AV_CODEC_ID_MP3: s->buf_ptr = s->buf + 4; break; - case CODEC_ID_MPEG1VIDEO: - case CODEC_ID_MPEG2VIDEO: + case AV_CODEC_ID_MPEG1VIDEO: + case AV_CODEC_ID_MPEG2VIDEO: break; - case CODEC_ID_MPEG2TS: + case AV_CODEC_ID_MPEG2TS: n = s->max_payload_size / TS_PACKET_SIZE; if (n < 1) n = 1; s->max_payload_size = n * TS_PACKET_SIZE; s->buf_ptr = s->buf; break; - case CODEC_ID_AAC: - s->read_buf_index = 0; + case AV_CODEC_ID_H264: + /* check for H.264 MP4 syntax */ + if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) { + s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1; + } + break; + case AV_CODEC_ID_VORBIS: + case AV_CODEC_ID_THEORA: + if (!s->max_frames_per_packet) s->max_frames_per_packet = 15; + s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15); + s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length + s->num_frames = 0; + goto defaultcase; + case AV_CODEC_ID_ADPCM_G722: + /* Due to a historical error, the clock rate for G722 in RTP is + * 8000, even if the sample rate is 16000. See RFC 3551. */ + avpriv_set_pts_info(st, 32, 1, 8000); + break; + case AV_CODEC_ID_OPUS: + if (st->codec->channels > 2) { + av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n"); + goto fail; + } + /* The opus RTP RFC says that all opus streams should use 48000 Hz + * as clock rate, since all opus sample rates can be expressed in + * this clock rate, and sample rate changes on the fly are supported. */ + avpriv_set_pts_info(st, 32, 1, 48000); + break; + case AV_CODEC_ID_ILBC: + if (st->codec->block_align != 38 && st->codec->block_align != 50) { + av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n"); + goto fail; + } + if (!s->max_frames_per_packet) + s->max_frames_per_packet = 1; + s->max_frames_per_packet = FFMIN(s->max_frames_per_packet, + s->max_payload_size / st->codec->block_align); + goto defaultcase; + case AV_CODEC_ID_AMR_NB: + case AV_CODEC_ID_AMR_WB: + if (!s->max_frames_per_packet) + s->max_frames_per_packet = 12; + if (st->codec->codec_id == AV_CODEC_ID_AMR_NB) + n = 31; + else + n = 61; + /* max_header_toc_size + the largest AMR payload must fit */ + if (1 + s->max_frames_per_packet + n > s->max_payload_size) { + av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n"); + goto fail; + } + if (st->codec->channels != 1) { + av_log(s1, AV_LOG_ERROR, "Only mono is supported\n"); + goto fail; + } + case AV_CODEC_ID_AAC: + s->num_frames = 0; default: - if (st->codec->codec_type == CODEC_TYPE_AUDIO) { - av_set_pts_info(st, 32, 1, st->codec->sample_rate); +defaultcase: + if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) { + avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate); } s->buf_ptr = s->buf; break; } return 0; + +fail: + av_freep(&s->buf); + return AVERROR(EINVAL); } /* send an rtcp sender report packet */ static void rtcp_send_sr(AVFormatContext *s1, int64_t ntp_time) { - RTPDemuxContext *s = s1->priv_data; + RTPMuxContext *s = s1->priv_data; uint32_t rtp_ts; -#if defined(DEBUG) - printf("RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); -#endif + av_dlog(s1, "RTCP: %02x %"PRIx64" %x\n", s->payload_type, ntp_time, s->timestamp); - if (s->first_rtcp_ntp_time == AV_NOPTS_VALUE) s->first_rtcp_ntp_time = ntp_time; s->last_rtcp_ntp_time = ntp_time; - rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, AV_TIME_BASE_Q, + rtp_ts = av_rescale_q(ntp_time - s->first_rtcp_ntp_time, (AVRational){1, 1000000}, s1->streams[0]->time_base) + s->base_timestamp; - put_byte(s1->pb, (RTP_VERSION << 6)); - put_byte(s1->pb, 200); - put_be16(s1->pb, 6); /* length in words - 1 */ - put_be32(s1->pb, s->ssrc); - put_be32(s1->pb, ntp_time / 1000000); - put_be32(s1->pb, ((ntp_time % 1000000) << 32) / 1000000); - put_be32(s1->pb, rtp_ts); - put_be32(s1->pb, s->packet_count); - put_be32(s1->pb, s->octet_count); - put_flush_packet(s1->pb); + avio_w8(s1->pb, (RTP_VERSION << 6)); + avio_w8(s1->pb, RTCP_SR); + avio_wb16(s1->pb, 6); /* length in words - 1 */ + avio_wb32(s1->pb, s->ssrc); + avio_wb64(s1->pb, NTP_TO_RTP_FORMAT(ntp_time)); + avio_wb32(s1->pb, rtp_ts); + avio_wb32(s1->pb, s->packet_count); + avio_wb32(s1->pb, s->octet_count); + + if (s->cname) { + int len = FFMIN(strlen(s->cname), 255); + avio_w8(s1->pb, (RTP_VERSION << 6) + 1); + avio_w8(s1->pb, RTCP_SDES); + avio_wb16(s1->pb, (7 + len + 3) / 4); /* length in words - 1 */ + + avio_wb32(s1->pb, s->ssrc); + avio_w8(s1->pb, 0x01); /* CNAME */ + avio_w8(s1->pb, len); + avio_write(s1->pb, s->cname, len); + avio_w8(s1->pb, 0); /* END */ + for (len = (7 + len) % 4; len % 4; len++) + avio_w8(s1->pb, 0); + } + + avio_flush(s1->pb); } /* send an rtp packet. sequence number is incremented, but the caller must update the timestamp itself */ void ff_rtp_send_data(AVFormatContext *s1, const uint8_t *buf1, int len, int m) { - RTPDemuxContext *s = s1->priv_data; + RTPMuxContext *s = s1->priv_data; -#ifdef DEBUG - printf("rtp_send_data size=%d\n", len); -#endif + av_dlog(s1, "rtp_send_data size=%d\n", len); /* build the RTP header */ - put_byte(s1->pb, (RTP_VERSION << 6)); - put_byte(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); - put_be16(s1->pb, s->seq); - put_be32(s1->pb, s->timestamp); - put_be32(s1->pb, s->ssrc); + avio_w8(s1->pb, (RTP_VERSION << 6)); + avio_w8(s1->pb, (s->payload_type & 0x7f) | ((m & 0x01) << 7)); + avio_wb16(s1->pb, s->seq); + avio_wb32(s1->pb, s->timestamp); + avio_wb32(s1->pb, s->ssrc); - put_buffer(s1->pb, buf1, len); - put_flush_packet(s1->pb); + avio_write(s1->pb, buf1, len); + avio_flush(s1->pb); - s->seq++; + s->seq = (s->seq + 1) & 0xffff; s->octet_count += len; s->packet_count++; } /* send an integer number of samples and compute time stamp and fill the rtp send buffer before sending. */ -static void rtp_send_samples(AVFormatContext *s1, - const uint8_t *buf1, int size, int sample_size) +static int rtp_send_samples(AVFormatContext *s1, + const uint8_t *buf1, int size, int sample_size_bits) { - RTPDemuxContext *s = s1->priv_data; + RTPMuxContext *s = s1->priv_data; int len, max_packet_size, n; + /* Calculate the number of bytes to get samples aligned on a byte border */ + int aligned_samples_size = sample_size_bits/av_gcd(sample_size_bits, 8); - max_packet_size = (s->max_payload_size / sample_size) * sample_size; - /* not needed, but who nows */ - if ((size % sample_size) != 0) - av_abort(); + max_packet_size = (s->max_payload_size / aligned_samples_size) * aligned_samples_size; + /* Not needed, but who knows. Don't check if samples aren't an even number of bytes. */ + if ((sample_size_bits % 8) == 0 && ((8 * size) % sample_size_bits) != 0) + return AVERROR(EINVAL); n = 0; while (size > 0) { s->buf_ptr = s->buf; @@ -187,18 +343,17 @@ static void rtp_send_samples(AVFormatContext *s1, s->buf_ptr += len; buf1 += len; size -= len; - s->timestamp = s->cur_timestamp + n / sample_size; + s->timestamp = s->cur_timestamp + n * 8 / sample_size_bits; ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 0); n += (s->buf_ptr - s->buf); } + return 0; } -/* NOTE: we suppose that exactly one frame is given as argument here */ -/* XXX: test it */ static void rtp_send_mpegaudio(AVFormatContext *s1, const uint8_t *buf1, int size) { - RTPDemuxContext *s = s1->priv_data; + RTPMuxContext *s = s1->priv_data; int len, count, max_packet_size; max_packet_size = s->max_payload_size; @@ -250,7 +405,7 @@ static void rtp_send_mpegaudio(AVFormatContext *s1, static void rtp_send_raw(AVFormatContext *s1, const uint8_t *buf1, int size) { - RTPDemuxContext *s = s1->priv_data; + RTPMuxContext *s = s1->priv_data; int len, max_packet_size; max_packet_size = s->max_payload_size; @@ -272,7 +427,7 @@ static void rtp_send_raw(AVFormatContext *s1, static void rtp_send_mpegts_raw(AVFormatContext *s1, const uint8_t *buf1, int size) { - RTPDemuxContext *s = s1->priv_data; + RTPMuxContext *s = s1->priv_data; int len, out_len; while (size >= TS_PACKET_SIZE) { @@ -292,76 +447,159 @@ static void rtp_send_mpegts_raw(AVFormatContext *s1, } } -/* write an RTP packet. 'buf1' must contain a single specific frame. */ +static int rtp_send_ilbc(AVFormatContext *s1, const uint8_t *buf, int size) +{ + RTPMuxContext *s = s1->priv_data; + AVStream *st = s1->streams[0]; + int frame_duration = av_get_audio_frame_duration(st->codec, 0); + int frame_size = st->codec->block_align; + int frames = size / frame_size; + + while (frames > 0) { + int n = FFMIN(s->max_frames_per_packet - s->num_frames, frames); + + if (!s->num_frames) { + s->buf_ptr = s->buf; + s->timestamp = s->cur_timestamp; + } + memcpy(s->buf_ptr, buf, n * frame_size); + frames -= n; + s->num_frames += n; + s->buf_ptr += n * frame_size; + buf += n * frame_size; + s->cur_timestamp += n * frame_duration; + + if (s->num_frames == s->max_frames_per_packet) { + ff_rtp_send_data(s1, s->buf, s->buf_ptr - s->buf, 1); + s->num_frames = 0; + } + } + return 0; +} + static int rtp_write_packet(AVFormatContext *s1, AVPacket *pkt) { - RTPDemuxContext *s = s1->priv_data; + RTPMuxContext *s = s1->priv_data; AVStream *st = s1->streams[0]; int rtcp_bytes; int size= pkt->size; - uint8_t *buf1= pkt->data; -#ifdef DEBUG - printf("%d: write len=%d\n", pkt->stream_index, size); -#endif + av_dlog(s1, "%d: write len=%d\n", pkt->stream_index, size); - /* XXX: mpeg pts hardcoded. RTCP send every 0.5 seconds */ rtcp_bytes = ((s->octet_count - s->last_octet_count) * RTCP_TX_RATIO_NUM) / RTCP_TX_RATIO_DEN; - if (s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && - (ntp_time() - s->last_rtcp_ntp_time > 5000000))) { - rtcp_send_sr(s1, ntp_time()); + if ((s->first_packet || ((rtcp_bytes >= RTCP_SR_SIZE) && + (ff_ntp_time() - s->last_rtcp_ntp_time > 5000000))) && + !(s->flags & FF_RTP_FLAG_SKIP_RTCP)) { + rtcp_send_sr(s1, ff_ntp_time()); s->last_octet_count = s->octet_count; s->first_packet = 0; } s->cur_timestamp = s->base_timestamp + pkt->pts; switch(st->codec->codec_id) { - case CODEC_ID_PCM_MULAW: - case CODEC_ID_PCM_ALAW: - case CODEC_ID_PCM_U8: - case CODEC_ID_PCM_S8: - rtp_send_samples(s1, buf1, size, 1 * st->codec->channels); + case AV_CODEC_ID_PCM_MULAW: + case AV_CODEC_ID_PCM_ALAW: + case AV_CODEC_ID_PCM_U8: + case AV_CODEC_ID_PCM_S8: + return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); + case AV_CODEC_ID_PCM_U16BE: + case AV_CODEC_ID_PCM_U16LE: + case AV_CODEC_ID_PCM_S16BE: + case AV_CODEC_ID_PCM_S16LE: + return rtp_send_samples(s1, pkt->data, size, 16 * st->codec->channels); + case AV_CODEC_ID_ADPCM_G722: + /* The actual sample size is half a byte per sample, but since the + * stream clock rate is 8000 Hz while the sample rate is 16000 Hz, + * the correct parameter for send_samples_bits is 8 bits per stream + * clock. */ + return rtp_send_samples(s1, pkt->data, size, 8 * st->codec->channels); + case AV_CODEC_ID_ADPCM_G726: + return rtp_send_samples(s1, pkt->data, size, + st->codec->bits_per_coded_sample * st->codec->channels); + case AV_CODEC_ID_MP2: + case AV_CODEC_ID_MP3: + rtp_send_mpegaudio(s1, pkt->data, size); + break; + case AV_CODEC_ID_MPEG1VIDEO: + case AV_CODEC_ID_MPEG2VIDEO: + ff_rtp_send_mpegvideo(s1, pkt->data, size); break; - case CODEC_ID_PCM_U16BE: - case CODEC_ID_PCM_U16LE: - case CODEC_ID_PCM_S16BE: - case CODEC_ID_PCM_S16LE: - rtp_send_samples(s1, buf1, size, 2 * st->codec->channels); + case AV_CODEC_ID_AAC: + if (s->flags & FF_RTP_FLAG_MP4A_LATM) + ff_rtp_send_latm(s1, pkt->data, size); + else + ff_rtp_send_aac(s1, pkt->data, size); break; - case CODEC_ID_MP2: - case CODEC_ID_MP3: - rtp_send_mpegaudio(s1, buf1, size); + case AV_CODEC_ID_AMR_NB: + case AV_CODEC_ID_AMR_WB: + ff_rtp_send_amr(s1, pkt->data, size); break; - case CODEC_ID_MPEG1VIDEO: - case CODEC_ID_MPEG2VIDEO: - ff_rtp_send_mpegvideo(s1, buf1, size); + case AV_CODEC_ID_MPEG2TS: + rtp_send_mpegts_raw(s1, pkt->data, size); break; - case CODEC_ID_AAC: - ff_rtp_send_aac(s1, buf1, size); + case AV_CODEC_ID_H264: + ff_rtp_send_h264(s1, pkt->data, size); + break; + case AV_CODEC_ID_H263: + if (s->flags & FF_RTP_FLAG_RFC2190) { + int mb_info_size = 0; + const uint8_t *mb_info = + av_packet_get_side_data(pkt, AV_PKT_DATA_H263_MB_INFO, + &mb_info_size); + ff_rtp_send_h263_rfc2190(s1, pkt->data, size, mb_info, mb_info_size); + break; + } + /* Fallthrough */ + case AV_CODEC_ID_H263P: + ff_rtp_send_h263(s1, pkt->data, size); break; - case CODEC_ID_MPEG2TS: - rtp_send_mpegts_raw(s1, buf1, size); + case AV_CODEC_ID_VORBIS: + case AV_CODEC_ID_THEORA: + ff_rtp_send_xiph(s1, pkt->data, size); break; - case CODEC_ID_H264: - ff_rtp_send_h264(s1, buf1, size); + case AV_CODEC_ID_VP8: + ff_rtp_send_vp8(s1, pkt->data, size); break; + case AV_CODEC_ID_ILBC: + rtp_send_ilbc(s1, pkt->data, size); + break; + case AV_CODEC_ID_MJPEG: + ff_rtp_send_jpeg(s1, pkt->data, size); + break; + case AV_CODEC_ID_OPUS: + if (size > s->max_payload_size) { + av_log(s1, AV_LOG_ERROR, + "Packet size %d too large for max RTP payload size %d\n", + size, s->max_payload_size); + return AVERROR(EINVAL); + } + /* Intentional fallthrough */ default: /* better than nothing : send the codec raw data */ - rtp_send_raw(s1, buf1, size); + rtp_send_raw(s1, pkt->data, size); break; } return 0; } -AVOutputFormat rtp_muxer = { - "rtp", - NULL_IF_CONFIG_SMALL("RTP output format"), - NULL, - NULL, - sizeof(RTPDemuxContext), - CODEC_ID_PCM_MULAW, - CODEC_ID_NONE, - rtp_write_header, - rtp_write_packet, +static int rtp_write_trailer(AVFormatContext *s1) +{ + RTPMuxContext *s = s1->priv_data; + + av_freep(&s->buf); + + return 0; +} + +AVOutputFormat ff_rtp_muxer = { + .name = "rtp", + .long_name = NULL_IF_CONFIG_SMALL("RTP output"), + .priv_data_size = sizeof(RTPMuxContext), + .audio_codec = AV_CODEC_ID_PCM_MULAW, + .video_codec = AV_CODEC_ID_MPEG4, + .write_header = rtp_write_header, + .write_packet = rtp_write_packet, + .write_trailer = rtp_write_trailer, + .priv_class = &rtp_muxer_class, };